Hi Nathan-
In short, I have a hard time believing that MTU issues are the underlying
cause for many (or even any) VoIP audio delivery problems
Only if it was encapsulated over some form of TCP.
VoIP audio streams other than PCMU-encoded ones, so perhaps it's
possible other codecs are different
It might be worth checking for EVS, which has a lot of SDP options.
We've seen some endpoints (handsets) that stop encoding because they
didn't understand SDP asks from the receiver. Basically bugs, they get
fixed over time, but since EVS is newer and still in adoption phase,
that time is stretched out.
-Jeff
Quoting Nathan Anderson via VoiceOps <[email protected]>:
Yes, hosts or routers-in-the-middle that don't send ICMP type 3 code
4, or which drop such a message sent by another host instead of
forwarding it, do make me upset.
But...
In this case we're talking about relatively narrow-band,
widely-compressed RTP audio. Admittedly I rarely deal with any VoIP
audio streams other than PCMU-encoded ones, so perhaps it's possible
other codecs are different (though I'd be surprised...timeliness of
delivery in a real-time application like this is far more important
than efficiency of packing the data into as few frames as possible),
but I personally have never seen an RTP frame that comes close to
approaching standard Ethernet MTU. The packets are typically more
like a couple hundred bytes large.
And of course being UDP, TCP MSS doesn't enter into the picture, either.
In short, I have a hard time believing that MTU issues are the
underlying cause for many (or even any) VoIP audio delivery
problems...but, as the meme goes, "change my mind"; heh.
-- Nathan
FROM: VoiceOps [mailto:[email protected]] ON BEHALF
OF Pinchas Neiman via VoiceOps
SENT: Sunday, March 10, 2024 7:29 AM
TO: Alex Balashov
CC: VoiceOps
SUBJECT: Re: [VoiceOps] One Way Audio - Frontier Comm (Los Angeles area)
I have (on a rural area DSL line) a desk phone registered
directly on line 1, and line 2 over the VPN, whenever someone on
line 1 tells me I couldn't hear you well, I am saying calling you
back with another line, every time they will respond immediately Ah.
Now your voice is much better.
TCP connections are also much more reliable over the VPN than direct.
I am using WG over UDP with MTU 80 bytes lower than the worst
case general MTU.
I digged through my issue, and found that some hops in my long
list of local hops (last mile/s) are very unreliable, and not
responding when they drop (crime #1) a big packet even if DF was set
(crime #2), so best for me was to have wireguard do the
fragmentation on my side, as well as signal to the TCP connections
to lower their MSS automatically.
In other cases a VPN will also be able to patch TCP issues
related to asymmetric routing, or firewall timeouts.
To be noted,
#1 VPN device CPU should be fast enough to do the packaging, as
there is usually no hardware assistance for the VPN prepackaging.. a
good gigabit router could easily become a source of latency when it
involves something more than passing/nating packets between ports
#2 having a VPN without adjusting the MTU (either manually or
automatically) will increase packet loss, this is the source of the
myth that VPN is a overhead for VOIP
My understanding in networking may be flawed but this is my
practical experience accumulated so far.
On Sat, Mar 9, 2024 at 4:00 PM Alex Balashov via VoiceOps
<[email protected]> wrote:
No, it's true, consider me appropriately humbled. I
underappreciated the nuance of this issue. I thought we were
talking about something closer to the physicality of networks, not
packet inspection/filtering/etc.
-- Alex
On 9 Mar 2024, at 08:11, James Cloos <[email protected]> wrote:
"AB" == Alex Balashov writes:
I don't trust last mile networks to reliably deliver SIP calls.
I usually end up putting them into VPNs, TLS, etc.
AB> VPNs and TLS make last-mile networks more reliable? :-)
on the vpn side, wireguard (which runs over udp) certainly does.
I imagine ipsec sometimes can, too. but wg is easier.
-JimC
--
James Cloos <[email protected]>
OpenPGP: https://jhcloos.com/0x997A9F17ED7DAEA6.asc
--
Alex Balashov
Principal Consultant
Evariste Systems LLC
Web: https://evaristesys.com
Tel: +1-706-510-6800
_______________________________________________
VoiceOps mailing list
[email protected]
https://puck.nether.net/mailman/listinfo/voiceops
PINCHAS S. NEIMAN
Software Engineer At ESEQ Technology Corp.
845.213.1229 #2
_______________________________________________
VoiceOps mailing list
[email protected]
https://puck.nether.net/mailman/listinfo/voiceops
